Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 27 +++++ libavfilter/Makefile | 1 + libavfilter/af_asoftclip.c | 217 +++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 246 insertions(+) create mode 100644 libavfilter/af_asoftclip.c diff --git a/doc/filters.texi b/doc/filters.texi index 4dd1a5de85..465eeb4732 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2104,6 +2104,33 @@ audio, the data is treated as if all the planes were concatenated. A list of Adler-32 checksums for each data plane. @end table +@section asoftclip +Apply audio soft clipping. + +Soft clipping is a type of distortion effect where the amplitude of a signal is saturated +along a smooth curve, rather than the abrupt shape of hard-clipping. + +This filter accepts the following options: + +@table @option +@item type +Set type of soft-clipping. + +It accepts the following values: +@table @option +@item tanh +@item atan +@item cubic +@item exp +@item alg +@item quintic +@item sin +@end table + +@item param +Set additional parameter which controls sigmoid function. +@end table + @anchor{astats} @section astats diff --git a/libavfilter/Makefile b/libavfilter/Makefile index fef6ec5c55..682df45ef5 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -80,6 +80,7 @@ OBJS-$(CONFIG_ASETRATE_FILTER) += af_asetrate.o OBJS-$(CONFIG_ASETTB_FILTER) += settb.o OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o OBJS-$(CONFIG_ASIDEDATA_FILTER) += f_sidedata.o +OBJS-$(CONFIG_ASOFTCLIP_FILTER) += af_asoftclip.o OBJS-$(CONFIG_ASPLIT_FILTER) += split.o OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o diff --git a/libavfilter/af_asoftclip.c b/libavfilter/af_asoftclip.c new file mode 100644 index 0000000000..239b1d6a6b --- /dev/null +++ b/libavfilter/af_asoftclip.c @@ -0,0 +1,217 @@ +/* + * Copyright (c) 2019 The FFmpeg Project + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +enum ASoftClipTypes { + ASC_TANH, + ASC_ATAN, + ASC_CUBIC, + ASC_EXP, + ASC_ALG, + ASC_QUINTIC, + ASC_SIN, + NB_TYPES, +}; + +typedef struct ASoftClipContext { + const AVClass *class; + + int type; + double param; + + void (*filter)(struct ASoftClipContext *s, void **dst, const void **src, + int nb_samples, int channels); +} ASoftClipContext; + +#define OFFSET(x) offsetof(ASoftClipContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption asoftclip_options[] = { + { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, NB_TYPES-1, A, "types" }, + { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" }, + { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" }, + { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" }, + { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" }, + { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" }, + { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" }, + { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" }, + { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(asoftclip); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +#define SQR(x) ((x) * (x)) + +static void filter_dblp(ASoftClipContext *s, + void **dptr, const void **sptr, + int nb_samples, int channels) +{ + double param = s->param; + int n, c; + + for (c = 0; c < channels; c++) { + const double *src = sptr[c]; + double *dst = dptr[c]; + + switch (s->type) { + case ASC_TANH: + for (n = 0; n < nb_samples; n++) { + dst[n] = tanh(src[n] * param); + } + break; + case ASC_ATAN: + for (n = 0; n < nb_samples; n++) + dst[n] = 2. / M_PI * atan(src[n] * param); + break; + case ASC_CUBIC: + for (n = 0; n < nb_samples; n++) { + if (FFABS(src[n]) >= 1.5) + dst[n] = FFSIGN(src[n]); + else + dst[n] = src[n] - 0.1481 * pow(src[n], 3.0f); + } + break; + case ASC_EXP: + for (n = 0; n < nb_samples; n++) + dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.; + break; + case ASC_ALG: + for (n = 0; n < nb_samples; n++) + dst[n] = src[n] / (sqrt(param + src[n] * src[n])); + break; + case ASC_QUINTIC: + for (n = 0; n < nb_samples; n++) { + if (FFABS(src[n]) >= 1.25) + dst[n] = FFSIGN(src[n]); + else + dst[n] = src[n] - 0.08192 * pow(src[n], 5.); + } + break; + case ASC_SIN: + for (n = 0; n < nb_samples; n++) { + if (FFABS(src[n]) >= M_PI_2) + dst[n] = FFSIGN(src[n]); + else + dst[n] = sin(src[n]); + } + break; + } + } +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + ASoftClipContext *s = ctx->priv; + + s->filter = filter_dblp; + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + ASoftClipContext *s = ctx->priv; + AVFrame *out; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(outlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + + s->filter(s, (void **)out->extended_data, (const void **)in->extended_data, + in->nb_samples, in->channels); + + if (out != in) + av_frame_free(&in); + + return ff_filter_frame(outlink, out); +} + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_asoftclip = { + .name = "asoftclip", + .description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."), + .query_formats = query_formats, + .priv_size = sizeof(ASoftClipContext), + .priv_class = &asoftclip_class, + .inputs = inputs, + .outputs = outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index c51ae0f3c7..4d3039d6ba 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -72,6 +72,7 @@ extern AVFilter ff_af_asetrate; extern AVFilter ff_af_asettb; extern AVFilter ff_af_ashowinfo; extern AVFilter ff_af_asidedata; +extern AVFilter ff_af_asoftclip; extern AVFilter ff_af_asplit; extern AVFilter ff_af_astats; extern AVFilter ff_af_astreamselect; -- 2.17.1